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Duration 21 hours
Course Outline
Part I: Introduction
- Introduction
- Background and rationale for adoption
- Classification of VoIP technologies and their evolution
- SIP – core principles and concepts
- SIP standardization frameworks (RFC 3261 and associated industry standards)
- System Architecture
- UA – User Agent components
- Standardized server roles: Registrar, Location, Proxy, and Redirect
- Application server integration
- Identity verification and addressing mechanisms
- The SIP architectural model
- Server Operations and Functionality
- Registration processes
- Server behavior in Proxy and Redirect modes
- Differences between stateless and stateful Proxy implementations
- Location server functionalities
- SRV records and DNS resolution
- uri/url/urn structures, ENUM, and NAPTR records
- SIP Signaling Protocols (including Instant Messaging & Presence – IMP extensions)
- Message structure and format
- Request types and handling
- Response types and processing
- Illustrative call cycle analysis
- Header definitions and parameter usage
- IMP operational models
- SDP (Session Description Protocol)
- Media description methods
- Standardized codec lists
- Rules for session negotiation
- Call Flows and SIP Signaling
- SIP session establishment – Reference to RFC 3261
- Detailed analysis of sample call scenarios
- Conferencing protocols and IP PBX integration
- Dynamics of changing media within a session
- Implementation of IMP features
- Routing of SIP Requests and Responses
- VIA header processing
- ROUTE and RECORD-ROUTE header functions
- Interoperability with PSTN
- SIP-T and SIP-I standards
- SIP early media and trunking configurations
- Signaling alignment between SIP and PSTN
- SIP Security Considerations
- Secure SIP, Secure RTP, and Secure RTCP protocols
- Common implementation strategies for Secure SIP
- Operational Challenges and Future Directions
- Traversal techniques for NAT and firewalls
- Quality of Service (QoS) management
- Integration of SIP and SDP within 3GPP IMS architectures
- Summary and Strategic Discussion
Part II: Practical Application
- SIP Implementation in LAN Environments: XLite SIP UA + Asterisk
- Establishing Asterisk accounts with basic dial plans
- Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and various SIP endpoints (Polycom, Gigaset, Yealink, Linphone)
- Processes for registration, call initiation, and reception
- Peer-to-peer (P2P) call testing with Linphone
- Analysis of SIP Signaling Using Wireshark
- Server configuration for monitoring
- Tracking SIP signaling and RTP media streams
- Packet-level analysis and specific call isolation
- Voice quality assessment: Jitter buffers, DTMF signaling retrieval (RFC 2833, INFO), and troubleshooting codecs and DTMF (transcoding, GSM issues, tone duplication)
- Integration of VoIP monitoring tools
- SDP, Instant Messaging, and Presence (IM&P)
- Review of SDP parameters and attributes
- Application of SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Practical exercises for IM&P using XLite and Linphone
- SIP Call Flow Dynamics
- SIP Registration via DNS
- Analysis of SIP SRV records
- Endpoint registration utilizing DNS-SRV
- Call Flows involving DNS
- Signaling analysis using Wireshark
- Diagnosing DNS timeouts and latency issues
- SIP Registration via DNS
- Managing SIP Trunks
- Setup of test SIP trunk connections
- Mitigation of threats (DOS, DDOS, fraud, CPS limits)
- SIP Security Posture
- Implementing SIP security with IPSec
- Enhancing security via Secure SIP protocols
- Assessing fraud risks in IP telephony
- Defense against DDOS and other attack vectors
- Deployment of SIP-Based VoIP Services
- Switch configuration standards
- SIP client setup and registration procedures
- Software Stack
- Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
- Linux CentOS operating system
- TDM2IP drivers
- Softphone applications (XLite, Linphone)
- Hardware Infrastructure
- Server systems
- TDM2IP cards/gateways
- Hardphone devices (Polycom, Gigaset, Yealink)
- Endpoint Configuration: Softphones and Hardphones
- Configuration Settings
- Codec selection
- Credential and server parameters (User/Password/SIP Server/Proxy/Ports)
- Operation and Signaling for:
- Three-way calling capabilities
- Call forwarding mechanisms
- Attendant call transfer
- Message Waiting Indication (MWI) and Busy Lamp Field (BLF)
- Yealink autoprovisioning protocols
- Vendor-specific operational constraints
- Configuration Settings
- SIP and Network Address Translation (NAT) Challenges
- NAT classification and structural analysis
- Application of STUN (Simple Traversal of UDP Through NATs)
- VoIP Call Quality Assurance and Troubleshooting
- Diagnosing connected calls with missing media
- Critical QoS indicators
- Latency, jitter, and play buffer sizing
- Voice quality metrics
- RTCP analysis for delay and jitter
- MOS calculation per ITU-T G.107 E-model
- Utilization of VoIP quality monitoring tools (Voipmonitor)
- Cloud-Hosted IP Telephony Solutions
- Conclusion and Addressing Participant-Submitted SIP and VoIP Issues
Testimonials (7)
Good contact with the trainer
Radziszewski - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
Detailed analysis of the provided traces
Krzysztof - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
examples presented
Slawomir - NBP
Course - SIP protocol in VoIP
The instructor patient to explain and show the step on each lab session.
CHEE WAI KAN - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Trainer can explain and give example for questions asked
SHAMSANI ABU BAKAR - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Very informative and detail explanation
Syahrifendi Lukmanilhakim - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
The tools and the contents of the slide.